Mobile devices frequently offer voice or audio communication systems so that users can communicate through a network with other users. Cellular telephones may offer voice communication using a cellular telephone network or the Internet. Other devices may use other networks to send and receive voice to remote devices. These systems use digital communication networks so that a downlink chain receives digital speech from the network, converts it to analog, and plays it through a speaker. An uplink chain receives speech through microphones, converts it to digital speech using an analog to digital converter (ADC) and sends the digital speech to the network. The digital speech from the ADC is in the form of discrete samples which are processed as samples and then packetized by being accumulated and attached to a packet header to be sent on the network. The same system is also used in some systems for audio and video recordings that are stored at the device.
The ADC in the uplink chain samples the speech to produce the digital speech. The sample rate and the number of bits per sample determine the quality of the digital conversion. For many systems an 8 kHz sample rate is used. If the analog audio is low pass filtered to include no audio frequencies higher than 4 kHz, then the Nyquist criterion is satisfied. Higher fidelity speech is currently being investigated for which higher audio frequencies are included in the digitized speech. The 4 kHz low pass filtered speech has been referred to as Narrowband Speech and 8 kHz low pass filtered speech is referred to as Wideband Speech. Super Wideband Speech uses a 16 kHz low pass filter. These higher frequency analog audio signals require higher sampling rates if the Nyquist criterion is to be satisfied.
In typical present day mobile audio and systems, almost all audio processing chains are driven by interrupts, typically from an AFE (Audio Front End). For each of these interrupts, the AFE periodically extracts a sample from the ADC of the microphone and feeds the sample through the uplink speech processing chain. This is followed by extracting a sample from the downlink speech processing chain to a loudspeaker or other interface. As a result, the sampling rate requirement of the ADC is directly related to the interrupt rate of the AFE. A higher speed ADC requires a higher interrupt rate at the audio scheduler. The audio scheduler schedules the various audio processing blocks in a typical mobile audio processing pipeline. Doubling or quadrupling the sampling rate requires that more computational resources are used for processing speech samples and frames.
Typically a real-valued speech signal being fed through a speech processing chain in the uplink direction should be sampled at twice its bandwidth. This can be derived according to Nyquist's sampling theorem. For Narrowband Speech, the bandwidth of the speech signal is restricted to 4 kHz in which case the sampling rate of the ADC is 8 kHz. This applies not only to the ADC, but also to the samples supplied to the DAC on the DL and to other interfaces. This provides reciprocity. For Wideband Speech, the bandwidth of speech is restricted to 8 kHz and the sampling rate of the ADC is 16 kHz. The sampling rate of the ADC or the DAC dictates the interrupt rate of the AFE.